Re: [asterisk-users] 3rd party app store

[Tilghman Lesher]

On Friday 17 September 2010 22:52:02 Dean Collins wrote:
> > Tilghman Lesher wrote:
> > On Friday 17 September 2010 16:53:58 Dean Collins wrote:
> > > > Tilghman Lesher wrote:
> > > > On Friday 17 September 2010 12:51:16 Dean Collins wrote:
> > > > > I recently came across this email that I wrote in May 2008 ..
> > > > >  http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.h
> > > > >tml
> > > > >
> > > > > It's such a shame that Digium manhandled the project away from the
> > > > > community only to then bury it and not allow it to proceed. I
> > > > > really wonder when I look at the Apple iphone development community
> > > > > as to where the 3rd party Asterisk development community could have
> > > > > been if Digium didn't kill this project.
> > > >
> > > > It's not buried.  You can find the link on asterisk.org, under
> > > > "Applications": http://www.asteriskexchange.com/
> > >
> > > Wow when did that happen?
> >
> > Shockingly, it happened two years ago, at Astricon, shortly after the
> > email that you referenced.  It was even part of a keynote address at
> > Astricon.  I'm not sure why you weren't aware of this, as a ton of
> > publicity went out surrounding it.  Perhaps you've just forgotten that it
> > existed in the interim?
>
> Any thoughts on why the lack of traffic?

No.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk sip attack

[bayardo . sanchez]

This week I was experiencing attacks sip log into my accounts were more than 
1,000 requests for records Sip accounts in less than an hour THROUGH deny  the 
ip of my router access list in cisco and my asterisk server to go through the 
iptables drop ip attacker is a way for an account with another ip can not log 
into my asterisk server to add some command in my sip.conf for deny register 
account sip in my asterisk? 
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Re: [asterisk-users] 3rd party app store

[Dean Collins]

Any thoughts on why the lack of traffic?

 
Cheers,
Dean
 
 

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> Sent: Friday, 17 September 2010 6:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 3rd party app store
> 
> On Friday 17 September 2010 16:53:58 Dean Collins wrote:
> > > -Original Message-
> > > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > > boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> > > Sent: Friday, 17 September 2010 4:03 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] 3rd party app store
> > >
> > > On Friday 17 September 2010 12:51:16 Dean Collins wrote:
> > > > I recently came across this email that I wrote in May 2008 ..
> > > >  http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html
> > > >
> > > > It's such a shame that Digium manhandled the project away from the
> > > > community only to then bury it and not allow it to proceed. I really
> > > > wonder when I look at the Apple iphone development community as to
> > > > where the 3rd party Asterisk development community could have been if
> > > > Digium didn't kill this project.
> > >
> > > It's not buried.  You can find the link on asterisk.org, under
> > > "Applications": http://www.asteriskexchange.com/
> >
> > Wow when did that happen?
> 
> Shockingly, it happened two years ago, at Astricon, shortly after the email
> that you referenced.  It was even part of a keynote address at Astricon.  I'm
> not sure why you weren't aware of this, as a ton of publicity went out
> surrounding it.  Perhaps you've just forgotten that it existed in the interim?
> 
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
> 
> --
> __
> ___
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[asterisk-users] externip/localnet

[dotnetdub]

Hi All,

Is it possible to specify more than 1 localnet? I know this is an odd
question. I have a customer that has multiple sites linked by VPN.

Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24

We want to allow some access to the public IP address at the main site. For
this to work I need to use the externip and localnet directive. If I do this
it rewrites the SDP with the external IP address of the main site on dialog
with the VPN'd sites.

This means that I can either have the VPN endpoints working or have people
accessing from outside..

Any work arounds?

Thanks
Brian
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Re: [asterisk-users] 3rd party app store

[Tilghman Lesher]

On Friday 17 September 2010 16:53:58 Dean Collins wrote:
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> > Sent: Friday, 17 September 2010 4:03 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] 3rd party app store
> >
> > On Friday 17 September 2010 12:51:16 Dean Collins wrote:
> > > I recently came across this email that I wrote in May 2008 ..
> > >  http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html
> > >
> > > It's such a shame that Digium manhandled the project away from the
> > > community only to then bury it and not allow it to proceed. I really
> > > wonder when I look at the Apple iphone development community as to
> > > where the 3rd party Asterisk development community could have been if
> > > Digium didn't kill this project.
> >
> > It's not buried.  You can find the link on asterisk.org, under
> > "Applications": http://www.asteriskexchange.com/
>
> Wow when did that happen?

Shockingly, it happened two years ago, at Astricon, shortly after the email
that you referenced.  It was even part of a keynote address at Astricon.  I'm
not sure why you weren't aware of this, as a ton of publicity went out
surrounding it.  Perhaps you've just forgotten that it existed in the interim?

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Registration attempts

[Zeeshan Zakaria]

It means that fail2ban is not configured correctly on your machine. For me
it works fine, and in fact lately these registration/hack attempts have gone
up significantly, thanks to cloud computing I guess.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-17 5:28 PM, "dave george"  wrote:

I am getting several hundred registration attempts on my aserterisk per
minute.  I have fail2ban installed but it's not stopping the attempts.  Any
suggestions.  Whatever they are using is changing the  userid on each
attempt.

Latest IP: 209.172.57.219

Thanks,
Dave


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Re: [asterisk-users] Initial Audio Cut off

[Ujjval Karihaloo]

What is the difference between this and the other option suggested below?

Just put in:
Answer()
Wait(1.5)




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns
Sent: Friday, September 17, 2010 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

Have you tried something like this
exten x => 1,Answer()
exten x => n,Wait(2)
exten x=> n,(whatever you are doing now)

Thanks,
Lyle J. McKarns
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of the Nexus Management Plc group of companies


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Friday, September 17, 2010 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

We have already tried that...but still there is say 1.5 sec delay but the 
actual Audio first 2-4 secs still get cut off..

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, September 17, 2010 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

Just put in:
Answer()
Wait(1.5)


On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo  
wrote:
> With some carriers the initial Audio (2-4 secs) seems to get cut off 
> when using a Auto Attendant or Conf Meetme.
>
> Is there any known remedies for that. Just want to know if others have 
> seen that esp. with Level 3.
>
>
>
> If Auto Attendant says - "Welcome to ABC bank"
>
> Caller only hears "Bank"
>
>
>
> --
> _
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Re: [asterisk-users] 3rd party app store

[Dean Collins]

Wow when did that happen?

How come here is no reviews/traffic 

 
Cheers,
Dean
 
 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> Sent: Friday, 17 September 2010 4:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 3rd party app store
> 
> On Friday 17 September 2010 12:51:16 Dean Collins wrote:
> > I recently came across this email that I wrote in May 2008 ..
> >  http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html
> >
> > It's such a shame that Digium manhandled the project away from the
> > community only to then bury it and not allow it to proceed. I really wonder
> > when I look at the Apple iphone development community as to where the 3rd
> > party Asterisk development community could have been if Digium didn't kill
> > this project.
> 
> It's not buried.  You can find the link on asterisk.org, under "Applications":
> http://www.asteriskexchange.com/
> 
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
> 
> --
> __
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Registration attempts

[Fred Posner]

I wrote a script to help with these here:

http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block

To each their own... there's 1000 ways of combatting this.

---fred
http://qxork.com





On Sep 17, 2010, at 5:18 PM, dave george wrote:

> I am getting several hundred registration attempts on my aserterisk per
> minute.  I have fail2ban installed but it's not stopping the attempts.  Any
> suggestions.  Whatever they are using is changing the  userid on each
> attempt.
> 
> Latest IP: 209.172.57.219
> 
> Thanks,
> Dave
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Registration attempts

[dave george]

I am getting several hundred registration attempts on my aserterisk per
minute.  I have fail2ban installed but it's not stopping the attempts.  Any
suggestions.  Whatever they are using is changing the  userid on each
attempt.

Latest IP: 209.172.57.219

Thanks,
Dave


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Re: [asterisk-users] Rotary phone on Asterisk

[Joel Maslak]

My understanding was that pulse dialing from a channel bank was "iffy", but
not pulse reception, so long as the channel bank properly reports on/off
hook state - that there is no real "pulse detection" in the channel bank,
simply on/off hook status (looking at some of my documentation, "real" D-2,
D-3, and D-4 channel banks all used the LSB in the 6th code word to indicate
on/off hook status).  Is this no longer correct?  I'm using ESF, although I
think D4 would work similarly for this function.

I will contact Rhino as well, though - just to cover my bases.  I was very
impressed with their support previously.

On Fri, Sep 17, 2010 at 12:56 PM, John Novack  wrote:

>
>
> Danny Nicholas wrote:
>
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Joel Maslak
> *Sent:* Friday, September 17, 2010 12:29 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Rotary phone on Asterisk
>
>
>
> I'm trying to use a couple of old Western Electric type 500 phones (desk
> model, rotary dial).  These phones work fine, as tested with telco lines
> (they dial, receiver/transmitter works fine, etc).
>
> I'm running Asterisk 1.6.2.11.
>
> I can't get them to dial through Asterisk.  They are connected to a Rhino
> channel bank which is connected to Asterisk via a Sangnoma card (T1 with
> echo cancellation).  Other phones (touch tone) work fine, as does any phone
> with a pulse/tone switch, even when these electronic phones are in "pulse"
> mode.
>
> I'm thinking that Asterisk is a bit too picky about the timing of the
> rotary dial pulses to handle a mechanical system.  Is there any way to
> correct this?
>
>
>
> Check this out
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing
>
>
>
> Better read the link
> That refers to the TDM400 card. Posted to voip-info quite a few years ago,
> I believe it MIGHT have made it into a recent release, or coming soon.
>
> For T1 though, I believe the channel bank might be the issue.
> We have several channel banks of various types with various T1 cards.
> Adtran 750's aren't supposed to work properly, according to Adtran, other
> companies do.
> Better ask Rhino first, as the pulse detection and timing is a channel bank
> issue.
>
> 500 sets are pretty good with pulse speed and make/break ratio.  The #9
> dials are even better than the #7 ones, though all are fairly stable.
> Older phones, WE or others, can be more difficult, and may need to be
> repaired.
>
>
> John Novack
>
> --
>
> Dog is my Co-pilot
>
>
> --
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Re: [asterisk-users] 3rd party app store

[Tilghman Lesher]

On Friday 17 September 2010 12:51:16 Dean Collins wrote:
> I recently came across this email that I wrote in May 2008 ..
>  http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html
>
> It's such a shame that Digium manhandled the project away from the
> community only to then bury it and not allow it to proceed. I really wonder
> when I look at the Apple iphone development community as to where the 3rd
> party Asterisk development community could have been if Digium didn't kill
> this project.

It's not buried.  You can find the link on asterisk.org, under "Applications":
http://www.asteriskexchange.com/

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Not able to join conference

[khalid touati]

Hi Guys,
Paul
you meant a debug file while the problem is happening, actually the thing is
i cannot even reproduce this issue, I'll keep trying though, but is there a
way to debug just Meetme app output?

On Fri, Sep 17, 2010 at 1:04 PM, Danny Nicholas  wrote:

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul
> Belanger
> Sent: Friday, September 17, 2010 11:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Not able to join conference
>
> On Fri, Sep 17, 2010 at 9:24 AM, khalid touati 
> wrote:
> > in the dialplan, that would be a big help if you guys can help diagnose
> the
> > issue.
> >
> A debug log of the actually problem will be more helpful.
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
> Or at least some CLI output, since there are only a few (hundred/thousand?)
> 1.2 users.
>
>
> --
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>



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Re: [asterisk-users] Rotary phone on Asterisk

[John Novack]

Danny Nicholas wrote:



*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Joel 
Maslak

*Sent:* Friday, September 17, 2010 12:29 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Rotary phone on Asterisk

I'm trying to use a couple of old Western Electric type 500 phones 
(desk model, rotary dial).  These phones work fine, as tested with 
telco lines (they dial, receiver/transmitter works fine, etc).


I'm running Asterisk 1.6.2.11.

I can't get them to dial through Asterisk.  They are connected to a 
Rhino channel bank which is connected to Asterisk via a Sangnoma card 
(T1 with echo cancellation).  Other phones (touch tone) work fine, as 
does any phone with a pulse/tone switch, even when these electronic 
phones are in "pulse" mode.


I'm thinking that Asterisk is a bit too picky about the timing of the 
rotary dial pulses to handle a mechanical system.  Is there any way to 
correct this?


Check this out

http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing


Better read the link
That refers to the TDM400 card. Posted to voip-info quite a few years 
ago, I believe it MIGHT have made it into a recent release, or coming soon.


For T1 though, I believe the channel bank might be the issue.
We have several channel banks of various types with various T1 cards. 
Adtran 750's aren't supposed to work properly, according to Adtran, 
other companies do.
Better ask Rhino first, as the pulse detection and timing is a channel 
bank issue.


500 sets are pretty good with pulse speed and make/break ratio.  The #9 
dials are even better than the #7 ones, though all are fairly stable.
Older phones, WE or others, can be more difficult, and may need to be 
repaired.



John Novack

--

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[asterisk-users] quick 1.8 question on console/dsp

[Jerry Geis]

In 1.4 I used alsa.conf and Dial(Console/Dsp)

In 1.8 this is not working (as I had it) . I know there is a new 
chan_console
I'd like to try both.

What is the correct Dial() for ALSA direct?
What is the correct Dial() for chan_console?

I "thought" if chan_alsa was loaded it would default to old behaviour
if chan_console was not loaded.

Thanks,

jerry

--
This is the error I see in 1.8.
 "Console/dsp") in new stack
[Sep 17 14:38:34] WARNING[4694]: channel.c:5278 ast_request: No channel 
type registered for 'Console'
[Sep 17 14:38:34] WARNING[4694]: app_dial.c:2031 dial_exec_full: Unable 
to create channel of type 'Console' (cause 66 - Channel not implemented)


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Re: [asterisk-users] Initial Audio Cut off

[Lyle McKarns]

Have you tried something like this
exten x => 1,Answer()
exten x => n,Wait(2)
exten x=> n,(whatever you are doing now)

Thanks,
Lyle J. McKarns
---
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n|m Nexus Management
4 Industrial Parkway
Suite 101
Brunswick, Maine 04011
 
Tel (USA)   : 1 207 319 1105
Tel (UK)  : 0207 100 4968
Fax: 1 207 725 8552
Nexus Management, Inc.│ Registered Office:  4 Industrial Parkway, Suite 101, 
Brunswick, Maine.  04011│Company No. 19891257D, Registered in Maine│ A member 
of the Nexus Management Plc group of companies


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Friday, September 17, 2010 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

We have already tried that...but still there is say 1.5 sec delay but the 
actual Audio first 2-4 secs still get cut off..

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, September 17, 2010 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

Just put in:
Answer()
Wait(1.5)


On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo  
wrote:
> With some carriers the initial Audio (2-4 secs) seems to get cut off 
> when using a Auto Attendant or Conf Meetme.
>
> Is there any known remedies for that. Just want to know if others have 
> seen that esp. with Level 3.
>
>
>
> If Auto Attendant says - "Welcome to ABC bank"
>
> Caller only hears "Bank"
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
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Re: [asterisk-users] Initial Audio Cut off

[Ujjval Karihaloo]

We have already tried that...but still there is say 1.5 sec delay but the 
actual Audio first 2-4 secs still get cut off..

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, September 17, 2010 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

Just put in:
Answer()
Wait(1.5)


On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo
 wrote:
> With some carriers the initial Audio (2-4 secs) seems to get cut off when
> using a Auto Attendant or Conf Meetme.
>
> Is there any known remedies for that. Just want to know if others have seen
> that esp. with Level 3.
>
>
>
> If Auto Attendant says - "Welcome to ABC bank"
>
> Caller only hears "Bank"
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Initial Audio Cut off

[Ujjval Karihaloo]

Is DAHDI the Analog /PRI card..or something.. We never use it..

Call is delivered over SIP from the carrier...and plays the standard WAV file 
in Asterisk...

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, September 17, 2010 9:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Initial Audio Cut off


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Friday, September 17, 2010 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Initial Audio Cut off

With some carriers the initial Audio (2-4 secs) seems to get cut off when using 
a Auto Attendant or Conf Meetme.
Is there any known remedies for that. Just want to know if others have seen 
that esp. with Level 3.

If Auto Attendant says - "Welcome to ABC bank"
Caller only hears "Bank"

Happens almost 100% of the time with a DAHDI connection (line supervision 
issue).
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[asterisk-users] 5-7 second delay in connecting outgoing FXO calls

[Frank Tarczynski]

  
  

  I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.

When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end.  There's enough delay
time that I hear an additional ring after the PSTN number has answered
the call.  I've had people hang-up since they don't hear anything when
answering.

If I try the exact same call using an IAX route, the call is connected
at my end just as soon as the PSTN number answers.

I don't have any connection delays for incoming FXO calls.  They are
connected as soon as I answer the calls.

Can anyone give me some pointers on where to start looking?

Frank

  


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Re: [asterisk-users] 3rd party app store

[Dean Collins]

I recently came across this email that I wrote in May 2008 ..  
http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html 

It's such a shame that Digium manhandled the project away from the community 
only to then bury it and not allow it to proceed. I really wonder when I look 
at the Apple iphone development community as to where the 3rd party Asterisk 
development community could have been if Digium didn't kill this project.


(for those of you not involved in Asterisk back in 2208 here is the audio of 
that conference call.
http://recordings.talkshoe.com/TC-22622/TS-109845.mp3?dl=1   )


Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).




asterisk-users] FW: Asterisk 3rd party developed commercial software sales 
licensing platform
Dean Collins Dean at cognation.net 
Mon May 5 06:24:48 CDT 2008 
* Previous message: [asterisk-users] MeetMeAdmin() working problem 
* Next message: [asterisk-users] FW: Asterisk 3rd party developed commercial 
software sales licensing platform 
* Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] 

Hi Randy,

As discussed on Friday the 9th of May I would like to host this weeks
Voip Users Conference Call.

The purpose of this call is to discuss the community's feelings about an
Asterisk 3rd party developed commercial software sales licensing
platform.

The plan is that some form of documented published schema be implemented
that will allow for 3rd party software developers to sell their software
applications using a common licensing model similar to the way G729
licenses are sold by Digium. 

Basically this discussion came about for a 3rd party ecosystem question
a few weeks ago when Cory Andrews from VoIP supply was on the Voip-Users
conference call.

I asked the question - how much of VoIP Supply revenue is product
hardware versus applications - he said we don't sell any services such
as ITSP hosted Asterisk so I replied that wasn't what I was thinking of
and gave the example of Snap Dialer which is a low cost (I paid $20 for
it) application which allows me to dial names from Outlook.

He said they didn't sell any applications like this at all but would
consider selling them if this was an opportunity presented to him.

I then talked about some of the consulting I did for Salesforce.com and
how they have built an entire ecosystem of third party applications all
built by other people apart from salesforce.com but utilizing the
documented API's and application security /licensing etc.

My comments were that although Asterisk should always remain a free open
source application that developers need to eat and pay rent as well.

If there was some common marketplace that developers could sell small -
low cost third party applications to the Asterisk community that Digium
had some type of overview/management control over who listed etc that
this would deliver a stream of revenue that would encourage further
application development.

The question I then posed to the group was if anyone knew how Digium
managed the sale and licensing of the G729 codes.
And if this was an open published standard that could it be used as the
basis for the Asterisk ecosystem license model.

Now I know it's not perfect and can be hacked but everything can be
hacked. The idea is to build apps cheap enough that it's not worth the
effort of hacking. If anyone has some alternative suggestions on how
apps should be licensed we'd like to hear them this Friday.

I know there were discussions in the early days of the Mexuar launch
about how they could license a single channel of the Mexuar Corraleta
application rather than the entire server license for $2000. The issue
always came down to how we could license it to 1/ a single channel
license. 2/ tied to a single machine and not transferable (currently the
Mexuar license is hard coded in the application to the servers IP
address).

I know for me personally although I have donated to numerous bounty
requests (I even tried to get one developed for video conferencing a few
years ago that was around the $10,000 range) I haven't seen the ongoing
continual development that would benefit the Asterisk community.

*    I personally would be more than happy to pay for 'the next
generation of FOP', it was a great application when launched but there
is a lot more it could be offering.

*    I'd also like to implement a far smarter 'user dashboard'
similar to what Druid are developing.

*    Now I no longer work for Mexuar and don't have access to it
anymore I'd also like to pay for a single channel Mexuar license rather
than using 'lesser quality' experiences by other solutions.

*    Drawing on my own now defunct project - is the Asterisk user
community now ready for centrally provided services such as the
'off-deck processing' like the Tellme Speech Recognition Service
http://www.voi

Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Jonas Kellens]

On 09/17/2010 06:00 PM, Mark Deneen wrote:

On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens
  wrote:
   

On 09/17/2010 05:29 PM, Mark Deneen wrote:
 

On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens
wrote:

   

warning: exec file is newer than core file.

 

Jonas,

I encourage you to read the output.  Did you run gdb with a core file
dumped from the old build?  You need to generate a new core dump with
the new executable.

Best Regards,
Mark Deneen


   

1. I have re-compiled asterisk 1.6.2 with dont_optimize
2. I have generated the restart/reload of asterisk that I expercience
3. I have built a backtrace with gdb from the resulting core.pid file

Don't know what I could have been doing wrong...

Jonas.
 

Jonas,

What is the timestamp on your asterisk binary and what is the
timestamp of the core file?

Also, you restarted asterisk after installing the dont_optimize binary?

The message suggests that your core file is older than your
executable, which should not be possible.

Best Regards,
Mark Deneen
   


Thank you for your feedback.

I have stopped asterisk (/sbin/service asterisk stop), then re-compiled 
and then restarted asterisk (/sbin/service asterisk start)


[r...@asterisk16 ~]# ls -l /usr/sbin/asterisk
-rwxr-xr-x 1 root root 15880306 *Sep 17 11:41* /usr/sbin/asterisk
[r...@asterisk16 ~]# ls -l 
/tmp/core.asterisk16.jocan.local-2010-09-17T10\:03\:15+0200
-rw--- 1 root root 16609280 *Sep 17 10:03* 
/tmp/core.asterisk16.jocan.local-2010-09-17T10:03:15+0200



This tells you are right...

Jonas.
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Re: [asterisk-users] Initial Audio Cut off

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, September 17, 2010 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off

Just put in:
Answer()
Wait(1.5)


On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo
 wrote:
> With some carriers the initial Audio (2-4 secs) seems to get cut off when
> using a Auto Attendant or Conf Meetme.
>
> Is there any known remedies for that. Just want to know if others have
seen
> that esp. with Level 3.
>
>
>
> If Auto Attendant says – “Welcome to ABC bank”
>
> Caller only hears “Bank”
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
This is a good answer;  a better one would be
Answer
Waitexten(1.5,m)

So that folks who don't have the line delay won't have 2 seconds of silence.


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Re: [asterisk-users] Initial Audio Cut off

[C F]

Just put in:
Answer()
Wait(1.5)


On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo
 wrote:
> With some carriers the initial Audio (2-4 secs) seems to get cut off when
> using a Auto Attendant or Conf Meetme.
>
> Is there any known remedies for that. Just want to know if others have seen
> that esp. with Level 3.
>
>
>
> If Auto Attendant says – “Welcome to ABC bank”
>
> Caller only hears “Bank”
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] do carriers detect unusual / unauthorized VoIP calling patterns?

[C F]

I have had where the Phone provider (this was a PRI) cut long distance
service to a box that was compromised till we called them to assure
them that the security holes where fixed.


On Fri, Sep 17, 2010 at 1:10 PM, Jeff Brower  wrote:
> All-
>
> Recently an Asterisk server we host was hacked and used to route some 
> unauthorized calls.  We have since improved our
> security measures, including installation of fail2ban.
>
> The interesting thing is the way in which this was discovered.  The 
> unauthorized calls were occurring intermittently
> last Thurs evening thru Sat morning.  On Sat morning, some of our employees 
> were attempting to log-in remotely to a
> company e-mail server and one found that his provider, Verizon, had blocked 
> the server static IP.
>
> My question:  do carriers build some type of "internal blacklist" if they 
> detect unusual VoIP calling patterns?  And
> possibly trade that between themselves, for example one carrier detects it, 
> and after some time other carriers are
> aware?  The carrier was used for the unauthorized calls is Tata... I'm 
> curious as to why Verizon (evidently) knew
> before Tata.
>
> -Jeff
>
> PS.  Interesting footnote:  upon learning of the Verizon block, one of our 
> employees drove to the lab and disconnected
> the VoIP subnet (with the Asterisk box), reset some routers, etc in an 
> attempt to get the company remote e-mail
> working again.  He didn't know it at the time, but in so doing, he cut off 
> the hackers "in mid call" (hehe) and saved
> a bunch of $$.
>
>
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Re: [asterisk-users] Rotary phone on Asterisk

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joel Maslak
Sent: Friday, September 17, 2010 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Rotary phone on Asterisk

 

I'm trying to use a couple of old Western Electric type 500 phones (desk
model, rotary dial).  These phones work fine, as tested with telco lines
(they dial, receiver/transmitter works fine, etc).

I'm running Asterisk 1.6.2.11.

I can't get them to dial through Asterisk.  They are connected to a Rhino
channel bank which is connected to Asterisk via a Sangnoma card (T1 with
echo cancellation).  Other phones (touch tone) work fine, as does any phone
with a pulse/tone switch, even when these electronic phones are in "pulse"
mode.

I'm thinking that Asterisk is a bit too picky about the timing of the rotary
dial pulses to handle a mechanical system.  Is there any way to correct
this?

 

Check this out

http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing

 

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[asterisk-users] Rotary phone on Asterisk

[Joel Maslak]

I'm trying to use a couple of old Western Electric type 500 phones (desk
model, rotary dial).  These phones work fine, as tested with telco lines
(they dial, receiver/transmitter works fine, etc).

I'm running Asterisk 1.6.2.11.

I can't get them to dial through Asterisk.  They are connected to a Rhino
channel bank which is connected to Asterisk via a Sangnoma card (T1 with
echo cancellation).  Other phones (touch tone) work fine, as does any phone
with a pulse/tone switch, even when these electronic phones are in "pulse"
mode.

I'm thinking that Asterisk is a bit too picky about the timing of the rotary
dial pulses to handle a mechanical system.  Is there any way to correct
this?
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[asterisk-users] CallerId: behavior changed between 1.4.25.1 and 1.4.36 with .call files

[Antonio Moragues]

Hi All,

We have a production system running 1.4.25.1 and yesterday we upgraded it to
1.4.36. Basically we use this system to generate scheduled calls via .call
files.

Sample .call file used:

Channel: local/11...@context-out
WaitTime: 30
CallerId: 3
Extension: 2
Context: context-out
Priority: 1

With the sample call file in 1.4.25.1 the behavior was:

1 - The asterisk box calls 1 with callerid 3
2 - When 1 answer the call asterisk calls 2 with
callerid 1

With the sample call file in 1.4.36 the behavior is:

1 - The asterisk box calls 1 with callerid 3
2 - When 1 answer the call asterisk calls 2 with
callerid 3

Basically now the callerid specified in the call file is used for the two
call legs and before was only used on the first call leg.

Anyone knows why the callerid behavior changed between this two versions and
if there is any way to have 1.4.36 actuate like 1.4.25.1?

Thanks for your help!

Regards,

Toni.
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Re: [asterisk-users] Not able to join conference

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Friday, September 17, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Not able to join conference

On Fri, Sep 17, 2010 at 9:24 AM, khalid touati 
wrote:
> in the dialplan, that would be a big help if you guys can help diagnose
the
> issue.
>
A debug log of the actually problem will be more helpful.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

Or at least some CLI output, since there are only a few (hundred/thousand?)
1.2 users.


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Re: [asterisk-users] Asterisk 1.8 and CEL logging

[Bryant Zimmerman]

Is there the ability in the Asterisk 1.8 CEL logging to log the SIP 
endpoint IP as weell as the medie enpoint's ID's?

Thanks
Bryant

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Re: [asterisk-users] Not able to join conference

[Paul Belanger]

On Fri, Sep 17, 2010 at 9:24 AM, khalid touati  wrote:
> in the dialplan, that would be a big help if you guys can help diagnose the
> issue.
>
A debug log of the actually problem will be more helpful.

-- 
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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[asterisk-users] do carriers detect unusual / unauthorized VoIP calling patterns?

[Jeff Brower]

All-

Recently an Asterisk server we host was hacked and used to route some 
unauthorized calls.  We have since improved our
security measures, including installation of fail2ban.

The interesting thing is the way in which this was discovered.  The 
unauthorized calls were occurring intermittently
last Thurs evening thru Sat morning.  On Sat morning, some of our employees 
were attempting to log-in remotely to a
company e-mail server and one found that his provider, Verizon, had blocked the 
server static IP.

My question:  do carriers build some type of "internal blacklist" if they 
detect unusual VoIP calling patterns?  And
possibly trade that between themselves, for example one carrier detects it, and 
after some time other carriers are
aware?  The carrier was used for the unauthorized calls is Tata... I'm curious 
as to why Verizon (evidently) knew
before Tata.

-Jeff

PS.  Interesting footnote:  upon learning of the Verizon block, one of our 
employees drove to the lab and disconnected
the VoIP subnet (with the Asterisk box), reset some routers, etc in an attempt 
to get the company remote e-mail
working again.  He didn't know it at the time, but in so doing, he cut off the 
hackers "in mid call" (hehe) and saved
a bunch of $$.


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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Mark Deneen]

On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens
 wrote:
> On 09/17/2010 05:29 PM, Mark Deneen wrote:
>> On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens  
>> wrote:
>>
>>> warning: exec file is newer than core file.
>>>
>> Jonas,
>>
>> I encourage you to read the output.  Did you run gdb with a core file
>> dumped from the old build?  You need to generate a new core dump with
>> the new executable.
>>
>> Best Regards,
>> Mark Deneen
>>
>>
>
> 1. I have re-compiled asterisk 1.6.2 with dont_optimize
> 2. I have generated the restart/reload of asterisk that I expercience
> 3. I have built a backtrace with gdb from the resulting core.pid file
>
> Don't know what I could have been doing wrong...
>
> Jonas.

Jonas,

What is the timestamp on your asterisk binary and what is the
timestamp of the core file?

Also, you restarted asterisk after installing the dont_optimize binary?

The message suggests that your core file is older than your
executable, which should not be possible.

Best Regards,
Mark Deneen

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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Jonas Kellens]

On 09/17/2010 05:29 PM, Mark Deneen wrote:
> On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens  
> wrote:
>
>> warning: exec file is newer than core file.
>>  
> Jonas,
>
> I encourage you to read the output.  Did you run gdb with a core file
> dumped from the old build?  You need to generate a new core dump with
> the new executable.
>
> Best Regards,
> Mark Deneen
>
>

1. I have re-compiled asterisk 1.6.2 with dont_optimize
2. I have generated the restart/reload of asterisk that I expercience
3. I have built a backtrace with gdb from the resulting core.pid file

Don't know what I could have been doing wrong...

Jonas.


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[asterisk-users] need help with IVR dialplan

[haloha]

Hi list


i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server


the call flow
IP phone ---INVITE 1001> opensips -> ASterisk INVITE
5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP
phone (5000)

my case  is:
1/ IP phone(5000) --->Opensips
2/ IVR number : 1001
3/ IP Phone calls 1001 to opensips --> asterisk, ASterisk will play IVR
4/ IP phone press 1, asterisk will Dial(SIP/to_opensips/5001,20)
5/ there are some cases when asterisk send call back to opensips
 5.1/ if extension 5001 does not exist, opensips send 404 message back
to asterisk, then asterisk wait 10s to hangup the IP phone 5000
 5.2/ if extension 5001is real, opensips send ring ring back to
asterisk, then 5001 does not want to answer call
 5.2.1/  the call  is time out - then asterisk wait 10s to hangup the IP
phone 5000
 5.2.2/ the call is cancel by 5001 - asterisk receives cancel then
asterisk wait 10s to hangup the IP phone 5000
 5.2.3/ the Phone 5001 is busy - asterisk receive busy then asterisk
wait 10s to hangup the IP phone 5000


how to i force asterisk hangup IP PHONE 5000 when asterisk receives time
out|Cancel|busy from opensips

Thank you
Ha`
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Re: [asterisk-users] Sangoma A108 PCIe V2.0

[Nyamul Hassan]

While this is too many "eggs" in one basket, but can be useful if you have
"too many" E(T)1s say equivalent to a STM1 (OC3) or more.  In that case, it
would be too many boxes at 8ports / box.

Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 per
box on the labs quite frequently, although mostly for load testing.

Can asterisk handle that much load (32 E1) on a box?  We have seen asterisk
crash quite frequently with 2 x 4 E1 load on one box (chan_ss7), especially
when the calls per second is above 10.

Regards
HASSAN


On Fri, Sep 17, 2010 at 20:56, John Novack wrote:

>
>
> Geraint Lee wrote:
>
> i suppose that depends on the number of eggs and baskets you have... but
> i'm guessing not many of either since you're considering using a desktop
> board for this...
>
> 24 T1 ports, if my math is correct.
> Lots of "eggs" for any PC, desktop or not!
> Lots of circuits/channels to go South with one machine failure
>
> I would want to spread the load around
> But then, it isn't my problem!
>
> Perhaps Sangoma will give similar advice?
>
> John Novack
>
>
>
>  but, email sangoma support, they will tell you.
>
> On 17 September 2010 12:47, John Novack wrote:
>
>>
>>
>> Anita Hall wrote:
>> > Hi
>> >
>> > Does Sangoma 8-port card A108 support PCIe version 2.0 ?
>> >
>> Ask Sangoma They are very helpful
>> > The card is here
>> >
>> http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
>> >
>> > And we want to use 3 such cards in this motherboard because it has 3
>> > PCIe slots of version 2.0
>> >
>> >
>> http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm
>> >
>> > Is this a good idea ? Do you have any experience with multiple A108
>> > with PCIe on the same motherboard that supports PCIe 2.0 ?
>> >
>> > Any comments will be helpful.
>> >
>> Lot of eggs in one basket!
>>
>> John Novack
>>
>> > Thanks,
>> > Anita Hall
>> > Simmortel Voice.
>>
>> --
>>
>> Dog is my Co-pilot
>>
>>
>> --
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>>
>
>
> --
>
> Dog is my Co-pilot
>
>
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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Mark Deneen]

On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens  wrote:
>
> warning: exec file is newer than core file.

Jonas,

I encourage you to read the output.  Did you run gdb with a core file
dumped from the old build?  You need to generate a new core dump with
the new executable.

Best Regards,
Mark Deneen

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[asterisk-users] ISDN BRI call disconnection issue

[Gopalakrishnan A.N]

Hi,

 I have a Netmod ISDN BRI router and from the router I have connected the
analog port in Asterisk via FXO card. Two analog lines I have connected to
asterisk machine. When both the lines are established, after 31 minutes the
call is automatically disconnected.

While checking the log it shows as busy tone is detected because of this
existing call is disconnected.

Did anybody faced this kind of issue. Also some assistance would be much
appreciated.

-- 
Thank you  with regards,
Gopalakrishnan A.N,
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Re: [asterisk-users] Initial Audio Cut off

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval
Karihaloo
Sent: Friday, September 17, 2010 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Initial Audio Cut off

 

With some carriers the initial Audio (2-4 secs) seems to get cut off when
using a Auto Attendant or Conf Meetme. 

Is there any known remedies for that. Just want to know if others have seen
that esp. with Level 3.

 

If Auto Attendant says - "Welcome to ABC bank"

Caller only hears "Bank"

 

Happens almost 100% of the time with a DAHDI connection (line supervision
issue).

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[asterisk-users] Initial Audio Cut off

[Ujjval Karihaloo]

With some carriers the initial Audio (2-4 secs) seems to get cut off when using 
a Auto Attendant or Conf Meetme.
Is there any known remedies for that. Just want to know if others have seen 
that esp. with Level 3.

If Auto Attendant says - "Welcome to ABC bank"
Caller only hears "Bank"

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Re: [asterisk-users] Sangoma A108 PCIe V2.0

[John Novack]

Geraint Lee wrote:
i suppose that depends on the number of eggs and baskets you have... 
but i'm guessing not many of either since you're considering using a 
desktop board for this...

24 T1 ports, if my math is correct.
Lots of "eggs" for any PC, desktop or not!
Lots of circuits/channels to go South with one machine failure

I would want to spread the load around
But then, it isn't my problem!

Perhaps Sangoma will give similar advice?

John Novack



but, email sangoma support, they will tell you.

On 17 September 2010 12:47, John Novack > wrote:




Anita Hall wrote:
> Hi
>
> Does Sangoma 8-port card A108 support PCIe version 2.0 ?
>
Ask Sangoma They are very helpful
> The card is here
>

http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
>
> And we want to use 3 such cards in this motherboard because it has 3
> PCIe slots of version 2.0
>
>

http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm
>
> Is this a good idea ? Do you have any experience with multiple A108
> with PCIe on the same motherboard that supports PCIe 2.0 ?
>
> Any comments will be helpful.
>
Lot of eggs in one basket!

John Novack

> Thanks,
> Anita Hall
> Simmortel Voice.

--

Dog is my Co-pilot


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--

Dog is my Co-pilot

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[asterisk-users] Not able to join conference

[khalid touati]

Hi All,
We are running to a weird problem, we're using asterisk 1.2 as a production
server (I'm wiling to move very soon to more recent version) and our problem
is when somebody try to join a conference he's told that he's the only one
in the conference but in fact there is some 3 or 5 or whatever people in
that same conference, after several tries he can/cannot enter the conference
and meet with the people already in, here is the lines corresponding to conf
in the dialplan, that would be a big help if you guys can help diagnose the
issue.

exten => 8080,1,Answer
exten => 8080,2,Wait,1
exten => 8080,3,MeetMe(|MDci)
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Re: [asterisk-users] Call restriction for particular extension

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
A.N
Sent: Friday, September 17, 2010 6:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call restriction for particular extension

 

Hi,

 

How to create dialplan restriction for particular extensions..

-- 
Thank you  with regards,
Gopalakrishnan A.N,

The "easiest" way to do this is "Ex-girlfriend" logic.  There are some good
examples online and in the Asterisk book.





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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

> Check the SIP debug and see what is going on. Alternatively you could turn 
> off 
the qualify option with qualify=no.

I'll take a look at the sip debug, but qualify needs to stay on, so thats not 
an option.


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Re: [asterisk-users] How to Understand a pri intense debug span X

[Danny Dias]

Any hints please?

I would appreciate your valuabl help

Thanks

2010/9/16 Danny Dias 

> Hello my friends,
>
> I would like to understand the output from "pri intense debug span X", the
> Telco always says that their side is OK, but i always receive alarms,
> loosing connection, take a look:
>
> [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1:
> Recovering
> [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Unable to disable echo
> cancellation on channel 1
> [...]
> [Sep 16 13:18:24] NOTICE[30363] chan_zap.c: PRI got event: No more alarm
> (5) on Primary D-channel of span 1
>
> So, i'm working with pri intense debug span X, but the output is quite
> difficult to understand, what should i check from this output? Where should
> i find some information in order to make a debug and talk to my telco with
> "power" :)
>
> Here is an example of pri intense debug span:
>
> < Supervisory frame:
> [Sep 16 13:59:25] < SAPI: 00  C/R: 0 EA: 0
> <  TEI: 000EA: 1
> [Sep 16 13:59:25] < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> < N(R): 024 P/F: 1
> < 0 bytes of data
> [Sep 16 13:59:25] Handling message for SAPI/TEI=0/0
> [Sep 16 13:59:25] -- ACKing all packets from 23 to (but not including) 24
>  [Sep 16 13:59:25] -- Since there was nothing left, stopping T200 counter
> [Sep 16 13:59:25] -- Stopping T203 counter since we got an ACK
> [Sep 16 13:59:25] -- Nothing left, starting T203 counter
>  [Sep 16 13:59:25] -- Got RR response to our frame
> [Sep 16 13:59:25] -- Restarting T203 counter
>
> Thanks in advance
>
> --
> Salu2
>
>
>


-- 
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Re: [asterisk-users] Realtime semi-colon

[Andrew Thomas]

I'd forgot about doing it that way (I use that for $).

Thanks for the memory jog :)

Cheers
Andy


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 16 September 2010 13:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime semi-colon


On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
> Does anyone know how to send * a semi-colon from a realtime database.

> I know that * uses the semi-colon as a 'seperator' - but I need to be 
> able to use one in a command.  I know I can use \; in the non-realtime

> configs, but this doesn't work in realtime.

in /etc/asterisk/extensions.conf

[globals]
SEMICOLON=\;

Then use ${SEMICOLON} in realitime Hacky, but it's what I'm using at
the moment..

S
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Re: [asterisk-users] Attended Transfer does not release channels

[Wolfgang Pichler]

2010/9/17 Olivier 

>
>
> 2010/9/17 Wolfgang Pichler 
>
> Hi all,
>>
>> i have the following setup
>>
>> PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk
>> 1.6.2.9 -> SIP -> agent
>>
>>
>> Does work quit fine - then agent does have the abibility to transfer a
>> call to a third party - the agent can initiate the transfer over a web
>> interface - it does generate a asterisk manager atxfer request...
>>
>> So agent does initiate transfer - call flow is
>>
>> agent -> SIP -> callcenter asterisk -> NEW call over IAX -> routing server
>> -> PSTN
>>
>> Then agent hangs up - so that the original caller and the new call will
>> get connected - and - it is working
>>
>> But - the call will not get released on the callcenter asterisk machine
>>
>> So the callflow after the transfer is
>>
>> Original call PSTN -> routing server -> callcenter asterisk -> routing
>> server -> PSTN
>>
>> But it should be
>>
>> Original call PTN -> routing server -> PSTN
>>
>> I have transfer = yes and mediaonly both tested on my connection routing
>> server to asterisk callcenter - does not help
>>
>> the iax peer beetween the both does have trunk=yes
>>
>> I do not get any error message (unable to transfer or something like this)
>>
>> I have done a full network dump of such a call - and i can see that
>> asterisk callcenter does not make any attempt to directly bridge the calls -
>> no TXREQ or something like that.
>>
>>
>>
>> So - why does it not try to directly bridge the both channels ?
>>
>
> see http://issues.asterisk.org/view.php?id=17999 and related bugs
>
I have taken a look at these bugs - but they don't seem to be related to my
problem - then transfer is working in my scenario - the problem is that the
call legs are not getting optimized out as it should be the case...

A calls B - B makes attended transfer to C -> B talks to C -> B hangs  up ->
asterisk should optimize out the call leg A -> B and B -> C to only A->C if
it is possible



>> I am using a local channel in the middle on asterisk callcenter - with /n
>> option - could this be the problem ?
>>
>> best regards,
>> Wolfgang
>>
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>
>
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Re: [asterisk-users] Sangoma A108 PCIe V2.0

[Geraint Lee]

i suppose that depends on the number of eggs and baskets you have... but i'm
guessing not many of either since you're considering using a desktop board
for this...

but, email sangoma support, they will tell you.

On 17 September 2010 12:47, John Novack wrote:

>
>
> Anita Hall wrote:
> > Hi
> >
> > Does Sangoma 8-port card A108 support PCIe version 2.0 ?
> >
> Ask Sangoma They are very helpful
> > The card is here
> >
> http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
> >
> > And we want to use 3 such cards in this motherboard because it has 3
> > PCIe slots of version 2.0
> >
> >
> http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm
> >
> > Is this a good idea ? Do you have any experience with multiple A108
> > with PCIe on the same motherboard that supports PCIe 2.0 ?
> >
> > Any comments will be helpful.
> >
> Lot of eggs in one basket!
>
> John Novack
>
> > Thanks,
> > Anita Hall
> > Simmortel Voice.
>
> --
>
> Dog is my Co-pilot
>
>
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Re: [asterisk-users] Sangoma A108 PCIe V2.0

[John Novack]

Anita Hall wrote:
> Hi
>
> Does Sangoma 8-port card A108 support PCIe version 2.0 ?
>
Ask Sangoma They are very helpful
> The card is here
> http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
>
> And we want to use 3 such cards in this motherboard because it has 3 
> PCIe slots of version 2.0
>
> http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm
>
> Is this a good idea ? Do you have any experience with multiple A108 
> with PCIe on the same motherboard that supports PCIe 2.0 ?
>
> Any comments will be helpful.
>
Lot of eggs in one basket!

John Novack

> Thanks,
> Anita Hall
> Simmortel Voice. 

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[asterisk-users] Determine busy state

[unserossi]

Hi all,

to be able to transfer calls I have set call-limit to 2 for all of my peers.
Now how can I determine if a peer is in busy state using the first line if I 
don't want to route a second call to it?

Thanks in advance,
Oliver
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[asterisk-users] Call restriction for particular extension

[Gopalakrishnan A.N]

Hi,

How to create dialplan restriction for particular extensions..

-- 
Thank you  with regards,
Gopalakrishnan A.N,
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[asterisk-users] Sangoma A108 PCIe V2.0

[Anita Hall]

Hi

Does Sangoma 8-port card A108 support PCIe version 2.0 ?

The card is here
http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html

And we want to use 3 such cards in this motherboard because it has 3 PCIe
slots of version 2.0

http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm

Is this a good idea ? Do you have any experience with multiple A108 with
PCIe on the same motherboard that supports PCIe 2.0 ?

Any comments will be helpful.

Thanks,
Anita Hall
Simmortel Voice.
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Re: [asterisk-users] Attended Transfer does not release channels

[Olivier]

2010/9/17 Wolfgang Pichler 

> Hi all,
>
> i have the following setup
>
> PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk
> 1.6.2.9 -> SIP -> agent
>
>
> Does work quit fine - then agent does have the abibility to transfer a call
> to a third party - the agent can initiate the transfer over a web interface
> - it does generate a asterisk manager atxfer request...
>
> So agent does initiate transfer - call flow is
>
> agent -> SIP -> callcenter asterisk -> NEW call over IAX -> routing server
> -> PSTN
>
> Then agent hangs up - so that the original caller and the new call will get
> connected - and - it is working
>
> But - the call will not get released on the callcenter asterisk machine
>
> So the callflow after the transfer is
>
> Original call PSTN -> routing server -> callcenter asterisk -> routing
> server -> PSTN
>
> But it should be
>
> Original call PTN -> routing server -> PSTN
>
> I have transfer = yes and mediaonly both tested on my connection routing
> server to asterisk callcenter - does not help
>
> the iax peer beetween the both does have trunk=yes
>
> I do not get any error message (unable to transfer or something like this)
>
> I have done a full network dump of such a call - and i can see that
> asterisk callcenter does not make any attempt to directly bridge the calls -
> no TXREQ or something like that.
>
>
>
> So - why does it not try to directly bridge the both channels ?
>

see http://issues.asterisk.org/view.php?id=17999 and related bugs

>
> I am using a local channel in the middle on asterisk callcenter - with /n
> option - could this be the problem ?
>
> best regards,
> Wolfgang
>
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Re: [asterisk-users] Issue with transfer (sip)

[Olivier]

2010/9/17 Benoit 

>
> Hi,
>
> I'm experiencing an issue with asterisk 1.6.2.10 & 12rc1,
> i'm not sure if it's to be expected or not, so here it is:
>
> When transferring call (blind-transfer) using asterisk feature key,
> things are working OK, however when using ZoIPer's transfer key
> (which is implemented with a "Refer-To" SIP message) the call is
> ended and the third party isn't even called.
>
> I've made a little bit of research and it seems to be because of a hangup
> handler in the current context. The handler is used for PRI hangup status
> analysis (and maybe mis-placed) and end with a "Hangup(${CAUSE})".
>
> But this handler isn't triggered when using the feature key.
>
> The question is who's fault is it (mine, zoiper's or asterisk's) ?
>

Asterisk's, it seems (see https://issues.asterisk.org/view.php?id=17999 and
related)


>
> thanks
>
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[asterisk-users] Sangoma A108 PCIe 2.0

[Anita Hall]

Hi

Does Sangoma 8-port card A108 support PCIe version 2.0 ?

The cards is here

And we want to use 3 such cards in this motherboard because it has 3 PCIe
slots of version 2.0

http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm

Is this a good idea ? Do you have any experience with multiple A108 with
PCIe on the same motherboard that supports PCIe 2.0 ?

Any comments will be helpful.

Thanks,
Anita Hall
Simmortel Voice.
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[asterisk-users] Issue with transfer (sip)

[Benoit]

Hi,

I'm experiencing an issue with asterisk 1.6.2.10 & 12rc1,
i'm not sure if it's to be expected or not, so here it is:

When transferring call (blind-transfer) using asterisk feature key,
things are working OK, however when using ZoIPer's transfer key
(which is implemented with a "Refer-To" SIP message) the call is
ended and the third party isn't even called.

I've made a little bit of research and it seems to be because of a hangup
handler in the current context. The handler is used for PRI hangup status
analysis (and maybe mis-placed) and end with a "Hangup(${CAUSE})".

But this handler isn't triggered when using the feature key.

The question is who's fault is it (mine, zoiper's or asterisk's) ?

thanks

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[asterisk-users] Attended Transfer does not release channels

[Wolfgang Pichler]

Hi all,

i have the following setup

PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk
1.6.2.9 -> SIP -> agent


Does work quit fine - then agent does have the abibility to transfer a call
to a third party - the agent can initiate the transfer over a web interface
- it does generate a asterisk manager atxfer request...

So agent does initiate transfer - call flow is

agent -> SIP -> callcenter asterisk -> NEW call over IAX -> routing server
-> PSTN

Then agent hangs up - so that the original caller and the new call will get
connected - and - it is working

But - the call will not get released on the callcenter asterisk machine

So the callflow after the transfer is

Original call PSTN -> routing server -> callcenter asterisk -> routing
server -> PSTN

But it should be

Original call PTN -> routing server -> PSTN

I have transfer = yes and mediaonly both tested on my connection routing
server to asterisk callcenter - does not help

the iax peer beetween the both does have trunk=yes

I do not get any error message (unable to transfer or something like this)

I have done a full network dump of such a call - and i can see that asterisk
callcenter does not make any attempt to directly bridge the calls - no TXREQ
or something like that.



So - why does it not try to directly bridge the both channels ?

I am using a local channel in the middle on asterisk callcenter - with /n
option - could this be the problem ?

best regards,
Wolfgang
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Re: [asterisk-users] changing from zap to DAHDI

[Tzafrir Cohen]

On Thu, Sep 16, 2010 at 10:03:09AM -0400, Jerry Geis wrote:
> Jerry Geis wrote:
> >>
> >> Somewhere on your system you have a modprobe install command that's
> >> running when the module is loaded.  Most likely it was installed on your
> >> system by
> >> http://svn.asterisk.org/view/zaptel/branches/1.4/build_tools/genmodconf?view=markup
> >>  
> >>
> >> when you installed zaptel.
> >>
> >> Do you have an /etc/conf.modules file?  What does 'grep -r ztconfig
> >> /etc/.' return?
> >>
> >>   
> >
> > Shaun,
> >
> > below is the results of the command.
> >
> > grep -r ztconfig /etc/.
> > grep: /etc/./httpd/run/asterisk.ctl: No such device or address
> > grep: /etc/./httpd/run/dbus/system_bus_socket: No such device or address
> > grep: /etc/./httpd/run/acpid.socket: No such device or address
> >
> > Jerry
> >
> >
> This is the new output (ztcfg is no longer mentioned) so I think that 
> issue is fixed.
> Now its:
> + initlog -q -c 'unload_module dahdi'
> execvp: No such file or directory

unload_module is a local function. Not an executable.

It should not be run using 'action'.

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[asterisk-users] Deadlock rendering sip useless

[Ingmar Steen]

Dear all,

We're experiencing some (what appear to be) deadlocks using
asterisk-1.4.35 (the problem also occurred on 1.4.24). Some of the
symptoms are that new SIP calls cannot be established and when running
"sip show channels" from the CLI, the CLI stops responding to any
further commands. The symptoms are in fact very similar to
https://issues.asterisk.org/view.php?id=15349. A possibly significant
difference is that once asterisk hangs and I try to stop or kill it,
asterisk will remain as a defunct process and I have to reboot the
machine to get things working again.

Two servers are involved, both running asterisk 1.4.35, both have a
TC400B card used for G729 transcoding and both are equipped with a
Sangoma A104dm card connected to 4x ISDN-30. Server A is the main server
(holds all phone registrations, does all the monitoring/recording, all
outgoing calls originate here and all incoming calls terminate here).
Server B acts as a SIP trunk in a neighboring country (connected by a
stable, dedicated and private link).

Every once in a while (about 2-4 times a week), either one of the
servers stops setting up new SIP channels, something we haven't
experienced before on other sites (only significant difference is that
the other sites use an older version of asterisk and lack a transcoder
card).

I've managed to get the output of "core show locks" (available here:
http://pastebin.com/S870RSi3), unfortunately, I haven't been able to
collect a backtrace yet.

Now down to my question: What I'd like to know is how to properly debug
this issue further.

On a side-note, there are a couple of problems with debugging this
issue: a) We've been unable to reproduce it locally. Actively trying to
reproduce it on the production machine is a no-go due to contracts with
the customers. And b) once the system does hang, we have very little
time to poke around (again, because of contracts), the system has to be
up and running as soon as possible.

Kind regards,
Ingmar Steen
Teleknowledge



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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Jonas Kellens]

On 09/16/2010 07:58 PM, Paul Belanger wrote:
> Please do not send me direct email, post them to the list for others
> to help.  Your backtrace is optimized ().  You
> need to reinstall asterisk with DONT_OPTIMIZE enabled, described in
> doc/backtrace.txt.
>

Hello,

I have compiled with DONT_OPTIMIZE.

When generating the backtrace, I get the following output now :

[r...@asterisk16 tmp]# ls
core.asterisk16.jocan.local-2010-09-17T10:03:15+0200
[r...@asterisk16 tmp]# gdb -se "/usr/sbin/asterisk" -ex "bt full" -ex 
"thread apply all bt" --batch -c 
/tmp/core.asterisk16.jocan.local-2010-09-17T10\:03\:15+0200 > 
/tmp/backtrace.txt

warning: exec file is newer than core file.

warning: .dynamic section for "/lib/libcrypt.so.1" is not at the 
expected address

warning: difference appears to be caused by prelink, adjusting expectations

warning: .dynamic section for "/lib/libnsl.so.1" is not at the expected 
address

warning: difference appears to be caused by prelink, adjusting expectations



Is the backtrace well generated and so can I post it ?! Or will I have 
to make adjustments and do it again ?

Jonas.

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